[asterisk-users] modules/cdr_odbc.so
Can anyone tell me if I can load the modules/cdr_odbc.so module without having to re compile my 1.4.20 production Asterisk? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with Nextone using H323
Any reason in particular why you don't use SIP between your Asterisk and NexTone? This is how I have ours connected and it works well. The only issue I've experienced is that some of the carriers that only support g729 AB have trouble with the dtmf tones from g729A, but this is not SIP specific. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo Salas M. Sent: Tuesday, June 24, 2008 11:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk with Nextone using H323 El mar, 24-06-2008 a las 12:20 -0300, Everton Goularth escribió: I`m using chan_ooh323 in my asterisk server. This is my ooh323.conf: Have you tried with chan_h323.so? I've one gateways that uses h.323 and works only with chan_h323.so . Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Avaya 4610sw
I have loaded the SIP firmware for an Avaya 4610sw IP phone and have successfully registered it to Asterisk BE and Asterisk 1.4.18. I am however experiencing two issues that I am hoping someone has already overcome. The first one is that the phone looses its registration from Asterisk every now and then. I found a tip that may work and am now testing which is to comment the line mailbox=(extension) from its sip.conf configuration. The second issue is that if I make a call and place the call on hold, when I pick up the line to resume the call, I hear no audio on neither the originating or destination phone. If I place the call on hold again, I can hear music on hold on the destination phone. image001.png___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Avaya 4610sw
I have an Avaya 4610SW IP phone which I have upgraded to SIP firmware. I have successfully registered this phone to Asterisk BE as well as Asterisk 1.4.18 Almost everything is working well. Except for two issues. One of the problems is that the phone looses registration every now and then ad I have to re register. I have found a tip for this which I am testing if it will work, which is to comment out line mailbox=(extension). The second and more serious issue is that when I place someone on hold, I am not able to resume the call. I can hear the music on hold on the destination phone and the music on hold stops when I try to pick up the line from the Avaya again, but there is no audio between the two phones. I can hang up and call again and I can hear both ways just fine. I would appreciate any input on this. Thank you Ed Nu ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Disable VAD on Polycom 330 or 301
Does anyone know an easy way to disable VAD on Polycom Phones? Thank you ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WARNING[26913]: channel.c:786 channel_find_locked: Avoided deadlock for '0x82d9668', 10 retries!
Is anyone familiar with this error message? WARNING[26913]: channel.c:786 channel_find_locked: Avoided deadlock for '0x82d9668', 10 retries! Why does it happen, and how can I prevent from happening. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanSpy issue
Good point, but the deal is that I have a remote call center with their own Nortel PBX. I get these calls from my DID provided via Zap and I send them VoIP to the gateway connected to the Nortel PBX. This is what I refer to my SIP trunk. When I specify Sip/SIPTRUNK(SIPTRUNK) is the name of the trunk. Asterisk only monitors one call at a time in the whole trunk, and you can Cycle through the calls by pressing *. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: Wednesday, September 26, 2007 8:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ChanSpy issue I am not an expert on chanspy, but it seems to me spying on the trunk would not work very well, would not you hear multiple conversations mixed if more than one extension were calling? Seems best to me to spy on an extension. YOu also can do a show channels to see who is talking to whom. on Wednesday 09/26/2007 Wai Wu([EMAIL PROTECTED]) wrote The parameter to Chanspy should be the whole or part of the channel name. I do not understand what you mean by sip trunk. It make perfect sense that you can hear both streams of voice when you use the phone's extension as Asterisk usually uses SIP/extension+xxx as the channel name of the call. -Original Message- From: [EMAIL PROTECTED] on behalf of Ed Nuñez Sent: Wed 9/26/2007 4:48 PM To: [EMAIL PROTECTED] Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] ChanSpy issue Hello list I am having an issue with Chanspy/SIP that I'm hoping someone has come across and resolved in the past. I am sending calls that come in TDM through T1 ZAP channels and go out to a SIP trunk. If I spy on the SIP channel, I can hear the person on the SIP side of the call just fine, but the person on the ZAP channel fades in and out. If I spy on the ZAP channel, and can hear both sides just fine, but I don't know who I am spying on since I have other calls coming in on the same T1. If I spy on a SIP extension instead of a SIP trunk, I hear both sides just fine. I am using a recent version of Asterisk 1.2 and I am using g729 licenses. This is the command I am using to spy. exten = 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4)) !DOCTYPE HTML PUBLIC -//W3C//DTD HTML 3.2//EN HTML HEAD META HTTP-EQUIV=Content-Type CONTENT=text/html; charset=iso-8859-1 META NAME=Generator CONTENT=MS Exchange Server version 6.5.7638.1 TITLERE: [asterisk-users] ChanSpy issue/TITLE /HEAD BODY !-- Converted from text/plain format -- PFONT SIZE=2The parameter to Chanspy should be the whole or part of the channel name. I do not understand what you mean by quot;sip trunkquot;. It make perfect sense that you can hear both streams of voice when you use the phone's extension as Asterisk usually uses quot;SIP/extension+xxxquot; as the channel name of the call.BR BR BR -Original Message-BR From: [EMAIL PROTECTED] on behalf of Ed NuñezBR Sent: Wed 9/26/2007 4:48 PMBR To: [EMAIL PROTECTED]BR Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'BR Subject: Re: [asterisk-users] ChanSpy issueBR BR BR BR Hello listBR BR BR BR I am having an issue with Chanspy/SIP that I'm hoping someone has comeBR across and resolved in the past.BR BR BR BR I am sending calls that come in TDM through T1 ZAP channels and go out to aBR SIP trunk.BR BR BR BR If I spy on the SIP channel, I can hear the person on the SIP side of theBR call just fine, but the person on the ZAP channel fades in and out.BR BR If I spy on the ZAP channel, and can hear both sides just fine, but I don'tBR know who I am spying on since I have other calls coming in on the same T1.BR BR BR BR If I spy on a SIP extension instead of a SIP trunk, I hear both sides justBR fine.BR BR BR BR I am using a recent version of Asterisk 1.2 and I am using g729 licenses.BR BR BR BR This is the command I am using to spy.BR BR BR BR exten =gt; 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4))BR BR BR BR BR BR BR BR BR BR BR /FONT /P /BODY /HTML___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided
Re: [asterisk-users] Ast_log
The Asterisk log file is normally located in /var/log/asterisk But you may want to read your asterisk.conf file to make sure the path in which your system store it. You will see something like this astlogdir = /var/log/asterisk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu Sent: Wednesday, September 26, 2007 3:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Ast_log Hi all, Anyone know where the asterisk log file is stored? I have some failed calls into my Asterisk box, and I just want to find out why those calls failed. Thnx. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanSpy issue
Hello list I am having an issue with Chanspy/SIP that Im hoping someone has come across and resolved in the past. I am sending calls that come in TDM through T1 ZAP channels and go out to a SIP trunk. If I spy on the SIP channel, I can hear the person on the SIP side of the call just fine, but the person on the ZAP channel fades in and out. If I spy on the ZAP channel, and can hear both sides just fine, but I dont know who I am spying on since I have other calls coming in on the same T1. If I spy on a SIP extension instead of a SIP trunk, I hear both sides just fine. I am using a recent version of Asterisk 1.2 and I am using g729 licenses. This is the command I am using to spy. exten = 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4)) image001.png___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue stats
Can anyone recommend a good commercial solution for queue statistics? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Music on hold 1.2
What is a good solution for playing music on hold on the 1.2 branch. I do not want to use mpg123 because last time I used it in a production server it caused many problems. The MPG123 process was taking about 60% of my Xeon CPU. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] kore dump
For anyone interested on the crashes I was experiencing when using ChanSpy from SIP extension to SIP extensions with the group option. For the last couple of days, Ive been monitoring from Zap extensions to SIP extensions, and the system has not crashed once. The problem only happens when I spy from SIP. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vadim Berezniker Sent: Tuesday, June 26, 2007 2:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] kore dump use the safe_asterisk script it will restart asterisk if it crashes and it enables core dumps (your core size limit is probably set to 0 when you start asterisk). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez Sent: Tuesday, June 26, 2007 2:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: [asterisk-users] kore dump I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server. My PBX has experienced several core dumps the last couple of days and I am not sure if this is whats causing it, but it always seems to happen when a particular extension on a grandstream phone uses ChanSpy SIP group. I have not been able to locate where the core dump file is being saved. I cant find it in my TMP directory. I would also like to know if Asterisk can be setup to automatically re start if there is a core dump. I was thinking of setting up a cron job to launch Asterisk every minute. If its running, no harm done, and if it crashes, the cron job will make sure that its started every 60 seconds. Any suggestions? Thank you Ed Nuñez ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] network routing
I have installed the Asterisk BE B.2.2 image file in a new server. I need to make network routing changes. However in their version of rPath (pound key) Digium has removed the netconfig command. I am able to manually add the route with Route add default gw xxx.xxx.xxx.xxx however when I reboot I lose my routing. Does anyone know which conf file I need to edit in order to make this routing change permanent? Thank you ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network routing
This allows me to edit the IP Address of the NIC card, but not edit my IP routing. Thanks From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ~Russell Sent: Thursday, June 28, 2007 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] network routing try to edit /etc/sysconfig/network-scripts/ifcfg-eth0 if u have eth0 if not try ifcfg-eth1 for eth1 On 6/29/07, Ed Nuñez [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have installed the Asterisk BE B.2.2 image file in a new server. I need to make network routing changes. However in their version of rPath (pound key) Digium has removed the netconfig command. I am able to manually add the route with Route add default gw xxx.xxx.xxx.xxx however when I reboot I lose my routing. Does anyone know which conf file I need to edit in order to make this routing change permanent? Thank you ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network routing
Thanks, that worked · I was using GATEWAYDEV=eth1 And that was not working. Thanks again From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ~Russell Sent: Thursday, June 28, 2007 3:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] network routing How many GW you need to add ? if it is one .. then add GATEWAY=xxx.xxx.xxx.xxx into /etc/sysconfig/network thanks Russell On 6/29/07, Ed Nuñez [EMAIL PROTECTED] wrote: I have installed the Asterisk BE B.2.2 image file in a new server. I need to make network routing changes. However in their version of rPath (pound key) Digium has removed the netconfig command. I am able to manually add the route with Route add default gw xxx.xxx.xxx.xxx however when I reboot I lose my routing. Does anyone know which conf file I need to edit in order to make this routing change permanent? Thank you ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] kore dump
What is a god Windows application to read core dump files?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of J. Oquendo
Sent: Tuesday, June 26, 2007 4:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] kore dump
Vadim Berezniker wrote:
use the safe_asterisk script
it will restart asterisk if it crashes and it enables core dumps (your
core size limit is probably set to 0 when you start asterisk).
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Ed
Nuñez
*Sent:* Tuesday, June 26, 2007 2:22 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
*Subject:* [asterisk-users] kore dump
I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server.
My PBX has experienced several core dumps the last couple of days and
I am not sure if this is whats causing it, but it always seems to
happen when a particular extension on a grandstream phone uses ChanSpy
SIP group.
I have not been able to locate where the core dump file is being
saved. I cant find it in my TMP directory.
I would also like to know if Asterisk can be setup to automatically re
start if there is a core dump. I was thinking of setting up a cron job
to launch Asterisk every minute. If its running, no harm done, and if
it crashes, the cron job will make sure that its started every 60
seconds.
Any suggestions?
Thank you
Ed Nuñez
--
--
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If that fails you could always try something like:
*/2 * * * * /bin/ps -C /usr/bin/asterisk || { /usr/bin/asterisk }
or so...
--
J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743
echo infiltrated.net|sed 's/^/sil@/g'
Wise men talk because they have something to say; fools, because they have
to say something. -- Plato
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Re: [asterisk-users] DTMF doesn't work between Asterisk and Cisco SIP Proxy
To configure the Cisco for RFC 2833 add the following line to the desired dial-peer dtmf-relay rtp-nte Hope this helps. Ed Nuñez -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Tuesday, June 26, 2007 11:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF doesn't work between Asterisk and Cisco SIP Proxy This is usually a Cisco issue. You need to set the Cisco to use RFC2833 DTMF. Check the Cisco docs. tracinet wrote: Jason, I am at least having similar issues with rfc2833 DTMF: http://bugs.digium.com/view.php?id=10058 On 6/20/07, Jason Ma [EMAIL PROTECTED] wrote: Hi buddies, I encountered DTMF issue when I tried to place call from x-lite to a sip conference serice,here is the diagram. X-liteAsterisk---Cisco SIP proxySIP Conference service The Call can be established,and I can hear from x-lite the prompt of the conference,but when I input any digits,nothing happened,the conference service did not recognize my input.At the same time,in the log of asterisk,I can find that asterisk recognized all the digitsI tried rfc2833,inband,info in the dtmfmode parameter,but did not work ,I'm not sure whether asterisk send the right dtmf to cisco proxy,how can I track that? I made another test,dialing from x-lite registered with Cisco proxy to voicemail service of Asterisk. x-liteCisco SIP proxyAsterisk---Voicemail service Both the call and dtmf worked fine,I can input my mailbox number and password and listen my voicemail.both rfc2933 and inband worked in this situation,but not info. My Asterisk is 1.4.4 with asterisk now,I did not configure dtmfmode in the section of xlite and the trunk to cisco proxy,just configure the dtmfmode in sip.conf. When I used rfc2833,I can see the log in asterisk as : [2007-06-19 16:01:40] DTMF[8925] channel.c: DTMF begin '2' received on SIP/-08269470 [2007-06-19 16:01:41] DTMF[8925] channel.c: DTMF end '2' received on SIP/-08269470, duration 160 ms [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF begin '1' received on SIP/-08269470 [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF end '1' received on SIP/-08269470, duration 140 ms and when I used inband,I can see : [2007-06-19 15:55:21] DTMF[8852] channel.c: DTMF end '2' received on SIP/-09d916c0, duration 0 ms [2007-06-19 15:55:22] DTMF[8852] channel.c: DTMF end '1' received on SIP/-09d916c0, duration 0 ms Is that right?Can I check what digits that asterisk sent out ? How can I track where is wrong with the dtmf?Did asterisk send dtmf to Cisco proxy correctly? I really have no idea about that.Please advise.Thank you very much ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] kore dump
I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server. My PBX has experienced several core dumps the last couple of days and I am not sure if this is what's causing it, but it always seems to happen when a particular extension on a grandstream phone uses ChanSpy SIP group. I have not been able to locate where the core dump file is being saved. I can't find it in my TMP directory. I would also like to know if Asterisk can be setup to automatically re start if there is a core dump. I was thinking of setting up a cron job to launch Asterisk every minute. If it's running, no harm done, and if it crashes, the cron job will make sure that it's started every 60 seconds. Any suggestions? Thank you Ed Nuñez ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4.5
I am seeing a peculiar message on my console screen on my new installation of Asterisk 1.4.5I would appreciate any comments. Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED] Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] inband DTMF for g729
I have a similar issue with Qwest SIP. They only support rfc2833 in g729AB, and Asterisk is only G729A. Sprint works fine for me. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: Friday, June 22, 2007 3:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] inband DTMF for g729 Sounds like you need a new SIP carrier. G.729 has a way of destroying inband DTMF tones. --- Matthew Fredrickson Software Engineer Digium, Inc. On Jun 22, 2007, at 1:20 PM, Gary Chen wrote: Does anybody know why Asterisk does not support inband DTMF for G.729? Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it for our Asterisk IVR system. Any suggestion to solve this problem? Gary ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanSpy SIP
For anyone experiencing the same problem, I was able to make SpyChan work on SIP extensions using the b and v options. exten = _**.,1,ChanSpy(IAX2/1654|bv(4)) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nunez Sent: Tuesday, June 19, 2007 8:05 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ChanSpy SIP Has anyone succesfully tried using ChanSpy on SIP channels with the latest Asterisk 1.4? I tried ChanSpy(SIP/5060) to monitor SIP extension 5060 and the console displays, Monitoring Sip/5060, but I don't hear anything. I am able to monitor Zap channels. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] g729
Oddly enough the call was being recorded. In any case in case anyone is
having the same problem, here is what did to get rid of the errors. I am
now using Monitor instead of MixMonitor as Jaswinder suggested.
Thanks
exten =
_1NXXNXX,1,Set(CALLFILENAME=/var/spool/asterisk/monitor/CONTINEX-${CALLE
RID:6}-${EXTEN}-${TIMESTAMP}-OUT)
exten = _1NXXNXX,2,Set(CDR(accountCode)=${CALLERID}-${TIMESTAMP})
exten = _1NXXNXX,3,Set(CDR(UserField)=${MONITOR_FILENAME})
exten = _1NXXNXX,4,Set(CALLERID(number)=14073844200)
exten = _1NXXNXX,5,Monitor(${CALLFILENAME}.wav49||mb)
exten = _1NXXNXX,6,Dial(SIP/[EMAIL PROTECTED])
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
Singh
Sent: Wednesday, June 06, 2007 11:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] g729
I think asterisk first converts audio stream to slin for recording to
a wav file . Since you are using hardware g729 transcoder i think this
is what is causing the problem . Is the calla actually being recorded
? I suggest that you use monitor application since it can directly
record g729 audio stream and run some cron script with sox mixing the
IN and OUT files in 1 file .
On 06/06/07, Ed Nuñez [EMAIL PROTECTED] wrote:
Yes
This is my extensions.conf entry.
exten = _1NXNXXX,1,Set(DYNAMIC_FEATURES=automon)
exten =
_1NXXNXX,2,Set(CALLFILENAME=/var/spool/asterisk/monitor/CONTINEX-${CALLE
RID}-${EXTEN}-${TIMESTAMP}-OUT)
exten =
_1NXXNXX,3,Set(TOUCH_MONITOR=CONTINEX-${CALLERID}-${EXTEN}-${TIMESTAMP}-
OUT)
exten = _1NXXNXX,4,Set(CDR(accountCode)=${CALLERID}-${TIMESTAMP})
exten = _1NXXNXX,5,Set(CDR(UserField)=${MONITOR_FILENAME})
exten = _1NXXNXX,6,Set(CALLERID(number)=14073844200)
exten = _1NXXNXX,7,MixMonitor(${CALLFILENAME}.wav49)
exten = _1NXXNXX,8,Dial(SIP/[EMAIL PROTECTED],,wW)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
Singh
Sent: Wednesday, June 06, 2007 4:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] g729
Are you trying to record the conversation as well ?
On 06/06/07, Ed Nuñez [EMAIL PROTECTED] wrote:
I installed a hardware g729 codec card in my asterisk, and I'm getting
the
following error when calling from a g729 sip extension to a SIP trunk
also
set to g729. The call goes through just fine, but these error messages
keep
flying by until I disconnect the call.
Any ideas?
ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin
failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
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RE: [asterisk-users] g729
Just wanted to update anyone interested in this issue.
If I monitor a g729 SIP channel using ChanSpy, I am getting the same error
as when I use MixMon.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
Sent: Thursday, June 07, 2007 12:14 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] g729
Oddly enough the call was being recorded. In any case in case anyone is
having the same problem, here is what did to get rid of the errors. I am
now using Monitor instead of MixMonitor as Jaswinder suggested.
Thanks
exten =
_1NXXNXX,1,Set(CALLFILENAME=/var/spool/asterisk/monitor/CONTINEX-${CALLE
RID:6}-${EXTEN}-${TIMESTAMP}-OUT)
exten = _1NXXNXX,2,Set(CDR(accountCode)=${CALLERID}-${TIMESTAMP})
exten = _1NXXNXX,3,Set(CDR(UserField)=${MONITOR_FILENAME})
exten = _1NXXNXX,4,Set(CALLERID(number)=14073844200)
exten = _1NXXNXX,5,Monitor(${CALLFILENAME}.wav49||mb)
exten = _1NXXNXX,6,Dial(SIP/[EMAIL PROTECTED])
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
Singh
Sent: Wednesday, June 06, 2007 11:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] g729
I think asterisk first converts audio stream to slin for recording to
a wav file . Since you are using hardware g729 transcoder i think this
is what is causing the problem . Is the calla actually being recorded
? I suggest that you use monitor application since it can directly
record g729 audio stream and run some cron script with sox mixing the
IN and OUT files in 1 file .
On 06/06/07, Ed Nuñez [EMAIL PROTECTED] wrote:
Yes
This is my extensions.conf entry.
exten = _1NXNXXX,1,Set(DYNAMIC_FEATURES=automon)
exten =
_1NXXNXX,2,Set(CALLFILENAME=/var/spool/asterisk/monitor/CONTINEX-${CALLE
RID}-${EXTEN}-${TIMESTAMP}-OUT)
exten =
_1NXXNXX,3,Set(TOUCH_MONITOR=CONTINEX-${CALLERID}-${EXTEN}-${TIMESTAMP}-
OUT)
exten = _1NXXNXX,4,Set(CDR(accountCode)=${CALLERID}-${TIMESTAMP})
exten = _1NXXNXX,5,Set(CDR(UserField)=${MONITOR_FILENAME})
exten = _1NXXNXX,6,Set(CALLERID(number)=14073844200)
exten = _1NXXNXX,7,MixMonitor(${CALLFILENAME}.wav49)
exten = _1NXXNXX,8,Dial(SIP/[EMAIL PROTECTED],,wW)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
Singh
Sent: Wednesday, June 06, 2007 4:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] g729
Are you trying to record the conversation as well ?
On 06/06/07, Ed Nuñez [EMAIL PROTECTED] wrote:
I installed a hardware g729 codec card in my asterisk, and I'm getting
the
following error when calling from a g729 sip extension to a SIP trunk
also
set to g729. The call goes through just fine, but these error messages
keep
flying by until I disconnect the call.
Any ideas?
ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin
failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
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RE: [asterisk-users] g729
Yes
This is my extensions.conf entry.
exten = _1NXNXXX,1,Set(DYNAMIC_FEATURES=automon)
exten =
_1NXXNXX,2,Set(CALLFILENAME=/var/spool/asterisk/monitor/CONTINEX-${CALLE
RID}-${EXTEN}-${TIMESTAMP}-OUT)
exten =
_1NXXNXX,3,Set(TOUCH_MONITOR=CONTINEX-${CALLERID}-${EXTEN}-${TIMESTAMP}-
OUT)
exten = _1NXXNXX,4,Set(CDR(accountCode)=${CALLERID}-${TIMESTAMP})
exten = _1NXXNXX,5,Set(CDR(UserField)=${MONITOR_FILENAME})
exten = _1NXXNXX,6,Set(CALLERID(number)=14073844200)
exten = _1NXXNXX,7,MixMonitor(${CALLFILENAME}.wav49)
exten = _1NXXNXX,8,Dial(SIP/[EMAIL PROTECTED],,wW)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
Singh
Sent: Wednesday, June 06, 2007 4:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] g729
Are you trying to record the conversation as well ?
On 06/06/07, Ed Nuñez [EMAIL PROTECTED] wrote:
I installed a hardware g729 codec card in my asterisk, and I'm getting the
following error when calling from a g729 sip extension to a SIP trunk also
set to g729. The call goes through just fine, but these error messages
keep
flying by until I disconnect the call.
Any ideas?
ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin
failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
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[asterisk-users] Voip-info.org
Is anyone else having trouble going into voip-info.org today? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] g729
I installed a hardware g729 codec card in my asterisk, and I'm getting the following error when calling from a g729 sip extension to a SIP trunk also set to g729. The call goes through just fine, but these error messages keep flying by until I disconnect the call. Any ideas? ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FW: autologoff
Is the autologoff function supported in Asterisk BE B.1-3? I have configured my agents.conf with a 5 second timeout, but the agents extension continues ringing until the call eventually goes to voicemail. Agents.conf [general] persistentagents=yes [agents] autologoff = 5 multiplelogin = no recordagencalls = yes monitor-join = yes createlink = yes updatecdr = yes musiconhold = default recordformat = wav49 savecallsin = /var/spool/asterisk/monitor/ agent = 1650,1650,Tareq agent = 1656,1656,Ed agent = 2000,2000,test agent agent = 1704,1704,Reload Test queues.conf [general] persistentmembers=yes [noi-cust-serv-spanish] strategy = leastrecent announce-frequency = 30 announce-holdtime = yes announce-round-seconds = 10 timeout=180 monitor-format=wav49 monitor-join=yes joinempty = strict leavewhenempty = strict musiconhold = default eventwhencalled = yes servicelevel=180 reportholdtime =yes maxlen=0; maximum ammount of calls waiting queue-youarenext = queue-youarenext; (You are now first in line.) queue-thereare = queue-thereare; (There are) queue-callswaiting = queue-callswaiting; (calls waiting.) queue-holdtime = queue-holdtime; (The current est. holdtime is) queue-minutes = queue-minutes ; (minutes.) queue-seconds = queue-seconds ; (seconds.) queue-thankyou = queue-thankyou; (Thank you for your patience.) queue-lessthan = queue-less-than ; (less than) queue-reporthold = queue-reporthold member = Agent/1656 autologoff - with this option you set for how long the phone has to ring with no answer, before the agent to be logged off. You have to set the maximum period of time in seconds. By default this option is set to 15 seconds. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FW: Re install
I had to re install the my Asterisk BE with the latest version, and when I try to load my g.729 codec license I do not see the folders in the path that they are described in the instructions given to us with the license or in your online documentation. I installed the disk 1 immage (rPath), and I am not able to perform the g.729 installation or registration. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] FW: Re install
I was able to fid the modules directoty, but when I run -r-x-- 1 root root 1288344 May 21 11:35 register /root/register I get the following error -bash: /root/register: cannot execute binary file I have changed the file attributes as you can see on the ls -l From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez Sent: Monday, May 21, 2007 11:25 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] FW: Re install I had to re install the my Asterisk BE with the latest version, and when I try to load my g.729 codec license I do not see the folders in the path that they are described in the instructions given to us with the license or in your online documentation. I installed the disk 1 immage (rPath), and I am not able to perform the g.729 installation or registration. image001.png___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AsteriskNow!
Does anyone know how to gain access directly to the configuration files in AsteriskNow? I have dual NICs and need to change the binding in the config file. I believe they blocked ssh2 access by default. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Autologoff
I am having an issue with the autologoff fuction in agents.conf I am running Asterisk BE and I am testing with agent 1656. I log in, and then make a call into the queue. The agent's phone rings, and if I answer it, all's fine/ I am trying to have this agent automatically be logged off if he does not answer the queue callback within 5 seconds, however the agents extension keeps ringing until the call eventually goes to the extension's voice mail, which I am also trying to avoid. Here is my agents.conf [general] persistentagents=yes [agents] autologoff=5 multiplelogin=no recordagencalls=yes monitor-join=yes createlink=yes updatecdr=yes musiconhold=default recordformat=wav49 urlprefix=http://xxx.xxx.xxx.xxx/calls/ savecallsin=/var/www/html/calls agent = 1650,1650, agent = 1656,1656,Ed Here is my queues.conf [general] persistentmembers=yes [noi-cust-serv-spanish] strategy = leastrecent announce-frequency = 90 announce-holdtime = yes announce-round-seconds = 10 timeout=180 monitor-format=wav49 monitor-join=yes joinwhenempty = strict leavewhenempty = yes musiconhold = default eventwhencalled = yes queue-youarenext = queue-youarenext; (You are now first in line.) queue-thereare = queue-thereare; (There are) queue-callswaiting = queue-callswaiting; (calls waiting.) queue-holdtime = queue-holdtime; (The current est. holdtime is) queue-minutes = queue-minutes ; (minutes.) queue-seconds = queue-seconds ; (seconds.) queue-thankyou = queue-thankyou; (Thank you for your patience.) queue-lessthan = queue-less-than ; (less than) member = Agent/1656 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Autologoff
I am having an issue with the autologoff fuction in agents.conf I am running Asterisk BE and I am testing with agent 1656. I log in, and then make a call into the queue. The agent's phone rings, and if I answer it, all's fine/ I am trying to have this agent automatically be logged off if he does not answer the queue callback within 5 seconds, however the agents extension keeps ringing until the call eventually goes to the extension's voice mail, which I am also trying to avoid. Here is my agents.conf [general] persistentagents=yes [agents] autologoff=5 multiplelogin=no recordagencalls=yes monitor-join=yes createlink=yes updatecdr=yes musiconhold=default recordformat=wav49 urlprefix=http://64.211.222.226/calls/ savecallsin=/var/www/html/calls agent = 1650,1650,Tareq Tujjar agent = 1656,1656,Ed Nuñez Here is my queues.conf [general] persistentmembers=yes [noi-cust-serv-spanish] strategy = leastrecent announce-frequency = 90 announce-holdtime = yes announce-round-seconds = 10 timeout=180 monitor-format=wav49 monitor-join=yes joinwhenempty = strict leavewhenempty = yes musiconhold = default eventwhencalled = yes queue-youarenext = queue-youarenext; (You are now first in line.) queue-thereare = queue-thereare; (There are) queue-callswaiting = queue-callswaiting; (calls waiting.) queue-holdtime = queue-holdtime; (The current est. holdtime is) queue-minutes = queue-minutes ; (minutes.) queue-seconds = queue-seconds ; (seconds.) queue-thankyou = queue-thankyou; (Thank you for your patience.) queue-lessthan = queue-less-than ; (less than) member = Agent/1656 image001.png___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Re Re: [asterisk-users] TC400B
The g729 licenses are US$10 a pop and you can buy them directly from www.Digium.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Wednesday, May 02, 2007 5:10 AM To: asterisk-users@lists.digium.com Subject: Re Re: [asterisk-users] TC400B Dear Andres; How much it cost the 4 licenses of G729 and from where I have to buy them? Also, what if I need to do IP Trunk between Asterisk and another IP PBX in another side (in case I need 30 ports for this IP Trunk, and I need to use G729 or G723 codec), then also I need to buy a license for this? How much? I was think that no licenses in Asterisk, now I see something new :) - Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Calls in ulaw, not gsm as desired
Reload will reload your sip.conf file! As well as iax.conf, extensions.conf, queues.conf, voicemail.conf, users.conf From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall Sent: Tuesday, May 01, 2007 2:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls in ulaw, not gsm as desired I was in the asterisk console and I typed reload. Is this not enough to reload the sip.conf file? Rob Andreas Sikkema wrote: However, even once I reloaded the extensions, its still only using ulaw. You didn't reload the sip config? Maybe that's your problem? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Confference function
I would like to know if anyone here knows the answer to the following question I need to implement the following conferencing feature for my agents. 1. Agent receives call from caller 2. Agent conferences a verification service 3. After finishing the verification, agent needs to drop third party (Verification service) and continue on the line with caller. My problem right now is being able to disconnect the third party and keeping the caller on the line. Would this be a function of Asterisk or the SIP / IAX phone? Any comments would be appreciated. Thank you Ed Nuñez ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk Pix firewalls
Don This may not be a solution to your question, but I would like to share that Ive been having one way audio issues when connecting point to sight to a PIX 515E using SIP. I changed to IAX and this is working perfectly now. It was paynless to configure IAX2, so you might want to consider it. Ed From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Don E. Wisdom Sent: Tuesday, April 24, 2007 8:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Pix firewalls Hi, I asked this last week but i didn't get any answer So i will elaborate on my question. I need to setup a pix 515 firewall (running 7.2.2 OS) to allow sip traffic thru it from a sip phone wherever i may be. The pix is where all my servers are colocated and i will need to connect thru it from softphones / hardphones wherever i happen to be traveling. I need help setting up the pix for inbound and outbound sip/iax traffic. Any help would be greatly appreciated. Thanks --Don ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP over VON
Hello all I would like to know if anyone here has had any experience trying to set up SIP or IAX over VPN. I am testing with Cisco VPN client and when I call the Asterisk server in my office I get one way audio. Thanks Ed Nunez ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Chanspy
In my asterisk, I have calls coming in on a Zap channel and going out SIP. My problem is that when I spy on the SIP channel, I hear the calling parting breaking in and out, and the called party sounds just fine (SIP). If I spy on the Zap channel , I hear both sides just fine. I am spying from my SIP extension. Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] MixMonitor and Queues
In queues.conf you must have the following under the queues you want to record.
monitor-format=wav49 ; you may also use wav or gsm formats
monitor-join=yes; if you have the latest sox installed,
thiswill join the in and out files into one.
In agents.conf
recordagencalls=yes
monitor-join = yes
recordformat=wav49
savecallsin=/var/www/html/calls ;this is the path where call will be
recorded.
That's all
If you want to change the file name place this in your extensions.conf on a
line prior to sending the call to the queue.
exten= 1097,4,Set(MONITOR_FILENAME=QUEUE-${CALLERID}-${TIMESTAMP})
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Moore
Sent: Wednesday, December 13, 2006 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] MixMonitor and Queues
Greetings, all.
I would like to record calls that are entered into queues and I'm not
quite sure how to do it. Here's how I'm currently set up:
- Call comes in and is placed into Queue #1 (which rings all phones for
15 sec).
- If call drops out of this queue, it is placed into Queue #2 (which
plays MoH until the call is picked up).
I've tinkered with MixMonitor and I have my queues set up, but I'm not
sure how to combine the two. Ideally, I'd like to only record once the
call comes out of queue (no point in recording hold music, unless I want
to hear people mumble about how lousy a company we are for placing them
on hold ;) )
On a semi-related note, is it possible to determine the extension that
pull the call out of queue before the call is bridged? The reason I ask
is that I'd like to put the receiving extension in the name of the file
that MixMonitor creates. If not, no biggie.
Recap:
Two queues. First rings for 15 seconds then drops into the second.
Second plays music on hold till the call is answered. I want to record
the call when it's pulled out of either queue using MixMonitor. Bonus
points if I can determine the answering extension before MixMonitor
starts (if possible).
Any help would be greatly appreciated.
Thanks,
Jay
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RE: [asterisk-users] MixMonitor and Queues
I've been trying to find where to download the Web Vmail application and instructions on how to install it for Asterisk BE. Any ideas? Thanks Ed Nuñez IT/Telecom Engineer 4037 Metric Drive Winter Park, FL (o) 407-384-4200 x 1656 (f) 407-384-4222 (c) 732-925-0730 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Moore Sent: Wednesday, December 13, 2006 10:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] MixMonitor and Queues Greetings, all. I would like to record calls that are entered into queues and I'm not quite sure how to do it. Here's how I'm currently set up: - Call comes in and is placed into Queue #1 (which rings all phones for 15 sec). - If call drops out of this queue, it is placed into Queue #2 (which plays MoH until the call is picked up). I've tinkered with MixMonitor and I have my queues set up, but I'm not sure how to combine the two. Ideally, I'd like to only record once the call comes out of queue (no point in recording hold music, unless I want to hear people mumble about how lousy a company we are for placing them on hold ;) ) On a semi-related note, is it possible to determine the extension that pull the call out of queue before the call is bridged? The reason I ask is that I'd like to put the receiving extension in the name of the file that MixMonitor creates. If not, no biggie. Recap: Two queues. First rings for 15 seconds then drops into the second. Second plays music on hold till the call is answered. I want to record the call when it's pulled out of either queue using MixMonitor. Bonus points if I can determine the answering extension before MixMonitor starts (if possible). Any help would be greatly appreciated. Thanks, Jay ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk manager
Your line number nine should also specify a file name to monitor to and the
format, like this
exten = 9,2,Monitor(from-${CALLEDID}-at-${TIMESTAMP},wav)
or better yet, use MixMon instead, because this will merge the two files into
just one. (both sides of the call)
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of nik600
Sent: Tuesday, December 12, 2006 5:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk manager
Hi
i am trying to record a call with
exten = 9,1,Answer
exten = 9,2,Monitor
exten = 9,3,Dial(SIP/200)
This will record the call, but asterisk generates 2 files in
/var/spool/asterisk/monitor/
-in.wav
-out.wav
Can i have only one file?
Can i customize the path where to save the files?
Thanks
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[asterisk-users] downloading asterisk GUI
This may be a Linux newby question, but here it goes. I was reading the instructions on downloading and installing Asterisk GUI, but I can't get this to work. svn checkout http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui What would be the equivalent command in CentOS 4? http://astrecipes.net/?n=217 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] downloading asterisk GUI
Thanks Ed Nuñez IT/Telecom Engineer 4037 Metric Drive Winter Park, FL (o) 407-384-4200 x 1656 (f) 407-384-4222 (c) 732-925-0730 _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mail list Sent: Friday, December 08, 2006 4:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] downloading asterisk GUI yum install subversion On 09/12/06, Kovar Petr [EMAIL PROTECTED] wrote: svn is application called subversion, you should download and install it first. - Original Message - From: Ed Nuñez mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com Sent: Friday, December 08, 2006 7:18 PM Subject: [asterisk-users] downloading asterisk GUI This may be a Linux newby question, but here it goes. I was reading the instructions on downloading and installing Asterisk GUI, but I can't get this to work. svn checkout http://svn.digium.com/svn/asterisk-gui/trunk http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui What would be the equivalent command in CentOS 4? http://astrecipes.net/?n=217 _ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users image001.gif Description: image001.gif ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue agent Monitor
Hello list.
Does anyone know if and how I can use in my context the following variable
found in the CDR field?
DSTCHANNEL
I am trying to make the answering agent part of the monitor file name, but it
is not working.
exten= 0072,1,Answer
exten= 0072,2,Ringing
exten= 0072,3,Wait(2)
exten= 0072,4,set(MONITORFILENAME=${DST_CHANNEL}${CALLERID}-${TIMESTAMP})
exten= 0072,5,Queue(NOC)
exten= 0072,6,Hangup
include = parkedcalls
#include users.conf
This is what I am getting for a file name.
4072493400-20061207-160632.wav
Caller - timestamp.wav
But I want to see
Agent(1656)-caller-timestamp.wav
Thank you
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
image001.gif
Description: image001.gif
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RE: [asterisk-users] queue agent Monitor
I just tried that and it doesn't work. This may be perhaps because the file
name needs to be defined before the call is sent to the queue.
When I saw you answer I thought it would work because it sounded very logical.
:-)
This is the macro I use to send the call to the extension
Just in case I put the line before and after the extension.
[macro-extensions]
exten = s,1,set(MONITOR_FILENAME=${EXTEN}-${CALLERID}-${TIMESTAMP})
exten = s,2,Dial(${ARG1}|30|t,,wW)
exten = s,3,set(MONITOR_FILENAME=${EXTEN}-${CALLERID}-${TIMESTAMP})
exten = s,4,Voicemail(u${ARG2})
exten = s,104,Voicemail(b${ARG2})
Ed Nuñez
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick
Sent: Thursday, December 07, 2006 3:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] queue agent Monitor
The queue application sends the call to an agent. Use the agent
extension's dialplan to set up the monitor, that way you will have the
actual agent extension.
On Thu, 2006-12-07 at 14:18 -0600, Ed Nuñez wrote:
Hello list.
Does anyone know if and how I can use in my context the following
variable found in the CDR field?
DSTCHANNEL
I am trying to make the answering agent part of the monitor file name,
but it is not working.
exten= 0072,1,Answer
exten= 0072,2,Ringing
exten= 0072,3,Wait(2)
exten= 0072,4,set(MONITORFILENAME=
${DST_CHANNEL}${CALLERID}-${TIMESTAMP})
exten= 0072,5,Queue(NOC)
exten= 0072,6,Hangup
include = parkedcalls
#include users.conf
This is what I am getting for a file name.
4072493400-20061207-160632.wav
Caller - timestamp.wav
But I want to see
Agent(1656)-caller-timestamp.wav
Thank you
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
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RE: [asterisk-users] problem with asterisk - calls where both sidescannot hear each other
If you use both the public and private interfaces for VoIP in the Asterisk
Server, make sure you don't specify one of them for the binding in sip.conf
Example
bindaddr=0.0.0.0
will allow SIP traffic on any of your interfaces.
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Singer Wang
Sent: Tuesday, December 05, 2006 4:23 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] problem with asterisk - calls where both sidescannot
hear each other
Hi,
I'm looking for some help with a problem in Asterisk (possibly), and I'm
confused as heck what is going on. I've updated to the latest Asterisk
version and the problem is still occur. My setup is as follows:
I've got Asterisk running on a high end Pentium-IV box running Linux
serving 5 calls, it is located in Canada. The calls come in via analog
lines through TDM400P cards to Asterisk box, which then converts it to
G729 channels to a call center in India over the Internet. Connection
between the Asterisk Server and the India call center is done via two
Cisco PIX501 devices, The call center in India is running 5 agents using
PolyCom phones, and we're using G729 to save bandwith. And yes, we
purchused 5 licenses of G729 codec.
We're using SIP and a ring all strategy, with the first agent that picks
up getting the call. The problem we're having is that about 5-10% calls
are not connecting properly. In that both sides can talk but do not hear
each other. Since we have recording in step s,5 (in the configuration
below), I can verify that it is happening. In these problematic calls,
both sides of the call talk but they cannot hear the other side at all.
I've gone through most of the documentation and spend hours on Google
search, does anyone have any idea what could be the problem? I'm willing
to provide more information if asked.
My extensions configuration is roughly the following:
[opened]
exten = s,1,SetVar(LOOP=1)
exten = s,2,Answer
exten = s,3,Wait(1)
exten = s,4,Background(open-hiq)
exten =
s,5,SetVar(MONITOR_FILENAME=/var/spool/asterisk/monitor/inbound/SUPPORT-${UNIQUEID})
exten = s,6,Queue(support3600)
exten = s,7,Voicemail(100|us)
exten = 1,1,Goto(opened,s,6)
exten = 500,1,Voicemail(500)
thanks,
Singer Wang
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RE: [asterisk-users] problem with asterisk - calls where both sidescannot hear each other
Singer
I would be interested to see the rest of your configuration pertaining to how
you are recording the calls. I am having trouble with this part.
Are you using monitor or MixMonitor from extensions.conf of are you using the
queues.conf or agents.conf monitor ?
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Singer Wang
Sent: Tuesday, December 05, 2006 4:23 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] problem with asterisk - calls where both sidescannot
hear each other
Hi,
I'm looking for some help with a problem in Asterisk (possibly), and I'm
confused as heck what is going on. I've updated to the latest Asterisk
version and the problem is still occur. My setup is as follows:
I've got Asterisk running on a high end Pentium-IV box running Linux
serving 5 calls, it is located in Canada. The calls come in via analog
lines through TDM400P cards to Asterisk box, which then converts it to
G729 channels to a call center in India over the Internet. Connection
between the Asterisk Server and the India call center is done via two
Cisco PIX501 devices, The call center in India is running 5 agents using
PolyCom phones, and we're using G729 to save bandwith. And yes, we
purchused 5 licenses of G729 codec.
We're using SIP and a ring all strategy, with the first agent that picks
up getting the call. The problem we're having is that about 5-10% calls
are not connecting properly. In that both sides can talk but do not hear
each other. Since we have recording in step s,5 (in the configuration
below), I can verify that it is happening. In these problematic calls,
both sides of the call talk but they cannot hear the other side at all.
I've gone through most of the documentation and spend hours on Google
search, does anyone have any idea what could be the problem? I'm willing
to provide more information if asked.
My extensions configuration is roughly the following:
[opened]
exten = s,1,SetVar(LOOP=1)
exten = s,2,Answer
exten = s,3,Wait(1)
exten = s,4,Background(open-hiq)
exten =
s,5,SetVar(MONITOR_FILENAME=/var/spool/asterisk/monitor/inbound/SUPPORT-${UNIQUEID})
exten = s,6,Queue(support3600)
exten = s,7,Voicemail(100|us)
exten = 1,1,Goto(opened,s,6)
exten = 500,1,Voicemail(500)
thanks,
Singer Wang
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[asterisk-users] Call recording with Asterisk BE
With Asterisk BE I am trying to record calls coming to a queue,. I am getting
the call to record, however the file name that the file saves to, is not the
correct one.
In my extensions.conf, I have the following entry to set the file name.
exten= 0072,1,Answer
exten= 0072,2,Ringing
exten= 0072,3,Wait(2)
exten= 0072,4,Set(AGENTFILENAME=${CALLERID(number)}-${TIMESTAMP}-${EXTEN:4})
exten= 0072,5,Monitor(wav,${AGENTFILENAME}),m
exten= 0072,6,Queue(NOC)
exten= 0072,7,Hangup
include = parkedcalls
#include users.conf
I have also tried
exten= 0072,4,Set(AGENTFILENAME=${CALLERID(number)}-${TIMESTAMP}-${EXTEN:4})
exten= 0072,5,Monitor(wav,${AGENTFILENAME},m)
but this is what I am getting in the file name.
agent-1656-1164843488-241-in.wav
agent-1656-1164843230-229-in.wav
In the Asterisk console the name appears correctly however.
-- Executing Set(SIP/1656-b7d10740,
AGENTFILENAME=1656-20061129-183350-) in new stack
-- Executing Monitor(SIP/1656-b7d10740, wav|1656-20061129-183350-|m) in
new stack
-- Executing Queue(SIP/1656-b7d10740, NOC) in new stack
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
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[asterisk-users] Call dropping
Can you please tell me what am I missing? If I don't make a choice before the last prompt starts playing the call is falling through. I want the menu to be replayed if a choice is not made, but the call gets disconnected and I see the following message in the Asterisk BE colsole. == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing GotoIfTime(SIP/1656-b7d1c868, 9:00-22:00|mon-sat|*|*?noi-main|s|1) in new stack -- Goto (noi-main,s,1) -- Executing Answer(SIP/1656-b7d1c868, ) in new stack -- Executing Wait(SIP/1656-b7d1c868, 2) in new stack -- Executing BackGround(SIP/1656-b7d1c868, main-welcome-open) in new stack -- Playing 'main-welcome-open' (language 'en') -- Executing BackGround(SIP/1656-b7d1c868, main-call-recorded) in new stack -- Playing 'main-call-recorded' (language 'en') -- Executing BackGround(SIP/1656-b7d1c868, main-select-option) in new stack -- Playing 'main-select-option' (language 'en') -- Executing BackGround(SIP/1656-b7d1c868, main-language-option) in new stack -- Playing 'main-language-option' (language 'en') == Auto fallthrough, channel 'SIP/1656-b7d1c868' status is 'UNKNOWN' This is what my config looks like. [noi-main] exten = s,1,Answer exten = s,2,Wait(2) exten = s,3,Background(main-welcome-open) exten = s,4,Background(main-call-recorded) exten = s,5,Background(main-select-option) exten = s,6,Background(main-language-option) exten = 1,1,GotoIfTime(9:00-22:00|mon-fri|*|*?noi-spanish,s,1) exten = 1,2,GotoIfTime(10:00-18:00|sat|*|*?noi-spanish,s,1) exten = 1,3,Goto(close,s,1) exten = 2,1,GotoIfTime(12:00-12:30|mon-sat|*|*?queue(orlando-local-arabic)) exten = 2,2,Goto(noi-arabic,s,1) exten = 3,1,Goto(noi-hindi,s,1) exten = 4,1,Goto(noi-english,s,1) exten = i,1,Playback(invalid) exten = i,2,Goto(s,3) exten = t,1,Goto(s,3) #include users.conf Ed Nuñez IT/Telecom Engineer 4037 Metric Drive Winter Park, FL (o) 407-384-4200 x 1656 (f) 407-384-4222 (c) 732-925-0730 _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Bishop Sent: Wednesday, November 29, 2006 5:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] g726 voice prompts Anyone know if it posible to make voice promps native g726 or g711 format? image001.gif Description: image001.gif ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk - Do Not Call List
I am trying to restrict calls to a Do Not Call List. These numbers are in a database of over 90 million records. I would like to know if anyone has already worked on this. Perhaps we can call a query from a script every time we make a call, and return a value that tells us if this number is in such list. Ed Nuñez IT/Telecom Engineer 4037 Metric Drive Winter Park, FL (o) 407-384-4200 x 1656 (f) 407-384-4222 (c) 732-925-0730 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Auto record a call?
This is how I'm able to record my outbound calls, hope this helps you.
exten = _407NXX,1,Set(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}-OUT)
exten = _407NXX,n,Monitor(wav,${CALLFILENAME},m)
exten = _407NXX,n,Dial(ZAP/g1/1${EXTEN:0})
exten = _407NXX,n,Congestion
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins
Sent: Wednesday, November 08, 2006 9:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Auto record a call?
I have a debugging scenario where I wish to record the entire call. The
call is establish via a .call file. I can't seem to get Monitor to do
anything. My dialplan looks like this:
[dialout]
exten = s,1,DigitTimeout,1
exten = s,n,ResponseTimeout,10
exten = s,n,Answer
exten = s,n,Monitor(wav,/tmp/test)
.
.
.
The file test.wav never shows up. Am I doing something wrong, or
possibly there is a better way to accomplish this?
Thanks,
MC
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[asterisk-users] Extension Spy
Is there a way to Specify an extension number to spy on, or monitor instead of specifying an agent or a SIP or ZAP channel? Ed Nuñez IT/Telecom Engineer 4037 Metric Drive Winter Park, FL (o) 407-384-4200 x 1656 (f) 407-384-4222 (c) 732-925-0730 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Extension Spy
It would be a SIP extension. Ed Nuñez IT/Telecom Engineer 4037 Metric Drive Winter Park, FL (o) 407-384-4200 x 1656 (f) 407-384-4222 (c) 732-925-0730 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Friday, November 03, 2006 3:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Extension Spy I don't understand... wouldn't your extension number be either a zap or sip channel? On 11/3/06, Ed Nuñez [EMAIL PROTECTED] wrote: Is there a way to Specify an extension number to spy on, or monitor instead of specifying an agent or a SIP or ZAP channel? Ed Nuñez IT/Telecom Engineer 4037 Metric Drive Winter Park , FL (o) 407-384-4200 x 1656 (f) 407-384-4222 (c) 732-925-0730 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk manager
I am trying to send commands to Asterisk manager via a telnet session. I am able to lo in and receive event logs from AMI, but when I try to issue commands I get an invalid/unknown command error. Here are some of the commands I am trying to send. Asterisk Call Manager/1.0 Action: login Username: xxx Secret: x Response: Success Message: Authentication accepted ACTION: Originate Channel: Local/1656 Exten: 1710 Priority: 1 Context: it Response: Error Message: Invalid/unknown command ACTION: Command command: show dialplan Response: Error Message: Invalid/unknown command Action: Originate Channel: Zap/g1/17329250730 Context: default Exten: 1656 Priority: 1 Callerid: 3125551212 Response: Error Message: Invalid/unknown command Here is how my manager.conf file looks [general] enabled = yes port = 5038 bindaddr = 0.0.0.0 displayconnects = yes [ami] secret = XXX permit=0.0.0.0/0.0.0.0 ;deny=0.0.0.0/0.0.0.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user Any help would be greatly appresiated Ed Nuñez IT/Telecom Engineer 4037 Metric Drive Winter Park, FL (o) 407-384-4200 x 1656 (f) 407-384-4222 (c) 732-925-0730 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk manager
Sorry, but I failed to mention that I am running Asterisk BE B 1-1 I am trying to send commands to Asterisk manager via a telnet session. I am able to lo in and receive event logs from AMI, but when I try to issue commands I get an invalid/unknown command error. Here are some of the commands I am trying to send. Asterisk Call Manager/1.0 Action: login Username: xxx Secret: x Response: Success Message: Authentication accepted ACTION: Originate Channel: Local/1656 Exten: 1710 Priority: 1 Context: it Response: Error Message: Invalid/unknown command ACTION: Command command: show dialplan Response: Error Message: Invalid/unknown command Action: Originate Channel: Zap/g1/17329250730 Context: default Exten: 1656 Priority: 1 Callerid: 3125551212 Response: Error Message: Invalid/unknown command Here is how my manager.conf file looks [general] enabled = yes port = 5038 bindaddr = 0.0.0.0 displayconnects = yes [ami] secret = XXX permit=0.0.0.0/0.0.0.0 ;deny=0.0.0.0/0.0.0.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user Any help would be greatly appresiated Ed Nuñez IT/Telecom Engineer 4037 Metric Drive Winter Park, FL (o) 407-384-4200 x 1656 (f) 407-384-4222 (c) 732-925-0730 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] auto recording extensions
I would like to know how to record all calls on a queue. Anu good sugestions? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] adding outbound prefix
Does anyone know how I can add a prefix to an outbound SIP call? I believe this would be done in extensions.conf, but am not sure how to go about it. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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From where can I download the collection of Asterisk Native Sounds? I tried the www.astlinux.com link, but I was not able to uncompress them because they seem to be corrupted. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
